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Incoming Obi calls to use VOIP provider's voicemail (call forking to voip.ms)

Started by dailyglen, December 02, 2011, 08:01:28 PM

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dailyglen

Hi,

My family/friends are getting Obi-s and to get free calls we call each other's Obi.  The problem is I use voip.ms for voicemail but calling directly to an Obi bypasses my voicemail.  I could setup a voicemail machine on the PH at home but I like having one voicemail box and voip.ms will email the voicemails and has a lot of other nice features.

After reading the admin guide I think I can fork calls coming in on the Obi trunk and ring both voip.ms and my ph with this setting for OBiTALK InboundCallRoute:

{(xxxxxxxxx):sp2($1>2151234567),ph},{ph}

This is untested but I believe it will take any 9 digit incoming number (assuming all obi numbers are 9 digit) and ring sp2 and ph1 simultaneously.  SP2 will keep the callers caller-id and phone my voip.ms number: 2151234567 (associated with sp1).  I think sp1 will be busy and then the voip.ms voicemail would pick up immediately which and not allow me to answer the call if I'm home.  I want the voip.ms to ring until there is no answer, then voicemail.  I think it would work if I had a separate voip.ms number but I don't want to pay $1/month for the DID. 

Has anyone tried something like this?  Any advice?

Thanks.

Stewart

This scheme would have several problems.  You don't say what's on SP2, but if it's another VoIP.ms sub-account, it won't pass caller ID as you expect.  If it's a different provider, you will probably be charged for the call.  Also, the call sent to VoIP.ms will be coming back to the OBi and conflict with the direct path to PH.

IMO, your friends' and family's OBi devices should be set up to call directly to your VoIP.ms account, instead of using OBiTALK.  You can use a speed dial coded with a SIP URI; the call will then not be charged.

infin8loop

Here's a Rube Goldberg approach.

This exercise will route unanswered calls on OBi110 LINE port to your voip.ms mailbox.  I can't think of a reason it wouldn't work on inbound OBiTALK calls on the OBi100/110 as well.  I don't know anyone with an OBi that could help me test.  Unfortunately inbound callerid number will be lost and calls in the mailbox will have your iNum on them.  

Given SP1: Google Voice and SP2: voip.ms
Google Voice is not used in this exercise, only voip.ms on SP2.  If voip.ms is on SP1 then change SP2 to SP1 in these instructions.

1. In your voip.ms account:
   a. Get a free iNum DID (inbound and outbound calls are free and no monthly fee)
   b. Go to DID Numbers -> CallerID Filtering and set up a new filter:
       Callerid: your new iNum number 8835100xxxxxxxx (replace the x's with your real iNum number)
       DID number to apply Filter: All   (just your iNum DID would probably work as well though I didn't test that)
       Set Routing and Failover Routing(Busy,Unreachable,No Answer) to the desired Mailbox
   c. Main Menu -> Account Setting -> Account Restrictions -> Allow International Calls : International Calls Enabled
   d. Main Menu -> Account Setting -> Account Restrictions -> Allow Calls to Countries :  iNum
       iNum is under Miscellaneous in the countries list. I suggest checking only countries you need. I only
       have iNum, Toll Free, and the United States selected.        

2. On the OBi110 (I configure locally):
   a.  Physical Interfaces > LINE Port > Calling Features >
                CallForwardOnNoAnswerEnable : Checked
                CallForwardOnNoAnswerNumber : sp2(0118835100xxxxxxxx)
                                                  (replace the x's with your real iNum. The 011 prefix is for International dialing.)      
                CallForwardOnNoAnswerRingCount : # of rings before going to VM

The above is tested and works.  Hopefully I've listed all parameters.

For unanswered OBiTALK inbound calls to voip.ms voice mail (untested):

3. On the OBi100/110:
   a.  Voice Services > OBiTALK Service > Calling Features >
                CallForwardOnNoAnswerEnable : Checked
                CallForwardOnNoAnswerNumber : sp2(0118835100xxxxxxxx)
                                                  (replace the x's with your real iNum. The 011 prefix is for International dialing.)      
                CallForwardOnNoAnswerRingCount : # of rings before going to VM

Why and how this works:  The unanswered call is routed to your iNum number provided by voip.ms.  Basically the OBi100/110 is calling itself (yourself?) but the CallerID filter at voip.ms traps incoming calls with your iNum as the callerid number and routes them immediately to voice mail.  Or something like that.  I need a drink.

Update:  It didn't occur to me until later that this same process works using a regular DID for a voip.ms account. My initial thought was that forwarding to a regular DID would incur a cost but since the forwarding is done through voip.ms to a voip.ms DID it's the same as calling another customer of voip.ms which is free. To implement the process this way, skip steps 1.a,c,d. In step 1.b use a regular voip.ms DID number instead of an iNum for Callerid of the filter.  In step 2.a and 3.a set CallForwardOnNoAnswerNumber : sp2(1231234567) where 1231234567 is the regular voip.ms DID. This requires voip.ms Main Menu -> Accounts Settings -> General -> CallerID Number to be the same as the regular DID the calls are forwarded to so the callerid number of the outbound (forwarded call) matches the DID in the voip.ms filter.  I left the voip.ms filter for the iNum and created another filter for the regular DID.  This way if in the future I want the flexibility of say forwarding calls from OBiTALK to one mailbox and calls from LINE port to another mailbox, I can simply forward one of them to the iNum and the other to the regular DID and change the filter rules at voip.ms to point the iNum filter to one mailbox and the regular DID filter to another mailbox. Or using the same CallForwardOnNoAnswer parameters in OBi's Voice Services > SP1 Service -> Calling Features I could set up Google Voice calls to forward unanswered calls to a voip.ms mailbox making sure CallForwardOnNoAnswerRingCount was set low enough to forward the call before the GV 25 second timeout routes the call to GV VM.
"This has not only been fun, it's been a major expense." - Gallagher

dailyglen

Hi,

infin8loop, thanks for the detailed response. The CallForwardOnNoAnswer* settings and call filtering did the magic but a I would suggest a few modifications:

1) From the Expert Config portal of the Obi, set the CallForwardOnNoAnswer* settings under "Voice Service -> Obitalk Service" (and not the LINE settings since the call is coming into Obitalk)
1.2) For CallForwardOnNoAnswerNumber I set it to my own DID associated with voip.ms.  On a incoming ObiTALK call, with no extra settings, the Obi will forward to voip.ms on no answer and voip.ms will ring the Obi back and I can receive the call.  I was hoping the Obi would ring busy when voip.ms sends the call back but it can complete the call (pretty amazing but I want it to go to voicemail instead).
2) In voip.ms portal, I used my own DID as the call filter and routed it to my voicemail.  This will prevent the undesired behavior in 1.2.  This has an added benefit that if I call my own number I get my voicemail which is useful setting since I use an IVR and can do other admin things as well as access voicemail.

So, in summary, my settings are:
1. In your voip.ms account:
  a. Go to DID Numbers -> CallerID Filtering and set up a new filter:
      Callerid: xxxxxxxxxx (my own DID number from voip.ms associated with my Obi110, eg I'm calling myself)
      DID number to apply Filter: All or my own DID
      Set Routing and Failover Routing(Busy,Unreachable,No Answer) to the desired Mailbox

2. On Obi 110 (I configure via Obitalk.com Expert configuration)
    a. Voice Service > ObiTALK Service > Calling Features >
           CallForwardOnNoAnswerEnable : Checked
           CallForwardOnNoAnswerNumber : xxxxxxxxxx [or, if not default route out] sp2(xxxxxxxxxx)
                                                   (replace the x's with your voip.ms DID as in step 1.a.)       
           CallForwardOnNoAnswerRingCount: 4 # of rings before going to VM


I have verified this and it works.  Thanks for the help!  Now my free ObiTALK calls will use my regular, voip.ms, voicemail!

If people have any improvements please post.  Is this worthy to go in the favorite tricks collection?

Cheers.

dailyglen

Hi Stewart,

Using a speed dial as a SIP URI sounds interesting if it is free.  Can you point me to more info or explain it further?

Thanks.

Stewart

Quote from: dailyglen on December 04, 2011, 08:04:45 PM
Using a speed dial as a SIP URI sounds interesting if it is free.  Can you point me to more info or explain it further?
Unfortunately, I've lost control of my OBi, so can't test this now.  However, I believe it should work to set up a Voice Gateway with AccessNumber of e.g. SP1(sip.voip.ms) and then set the speed dial value to e.g. VG1(yourDID).

If the above doesn't work, you could try iNum.  If you don't already have one, get a free iNum DID on your VoIP.ms account, and send the call either via the remote OBi's provider (if they offer free calling to iNum), or via a gateway to sip.inum.net .

(I assume that this is only of academic interest, as it appears that infin8loop already solved your problem.)

RonR

If you have any SIP Provider configured on SP1 and/or SP2, you can make SIP URI calls from Speed Dials:

User Settings -> Speed Dials -> 10 : SPx(883510000000091@sip.inum.net)

where SPx is SP1 or SP2 (must be configured for SIP).

dailyglen

Hi RonR,

Thanks for the tip.  Can I put SPx(883510000000091@sip.inum.net) directly in the CallForwardOnNoAnswerNumber field?  Ideally, I want to use:

SPx($1 > 883510000000091@sip.inum.net)

What I am trying to do is make an extension in voip.ms which is basically a SIP URI so that I can forward unanswered calls to it which is routed directly to my voicemail.  My problem with my current setup (above) is I have the wrong caller ID for the person who leaves the voicemail.  So I want to spoof the caller ID to be that of the original caller.  If I spoof the callerid and forward to my DID then I can't filter it to go directly to voicemail.  Perhaps I need to use an inum for this but then this inum is useless except for leaving voicemail.

Thanks.

RonR

dailyglen,

Much of this stuff is undocumented and you just have to try it to see if it works.

I suspect SPx(883510000000091@sip.inum.net) is valid in a CallForwardOnWhatever field, but I doubt that $1 is.  $1 is only documented in conjunction with rule processing.  Also, $1 didn't work correctly in earlier firmware versions, but it's not clear if it was a bug or a result of the intentional limiting of CallerID passing between trunks other than the OBiTALK trunk.

infin8loop


Does anyone have any ideas what would cause the following speed dials (10 and 12) to fail:

SP2=voip.ms (SIP)  xxxxxxxx=my iNum

10: SP2(8835100xxxxxxxx@sip.inum.net) fails with End Call (500 Service Unavailable)
   
11: SP2(0118835100xxxxxxxx)  immediately answered by Voice Mail at voip.ms per a filter rule at voip.ms on inbound callerid number 8835100xxxxxxxx (calling myself) as expected
   
12: SP2(883510000000091@sip.inum.net)  call connects but just dead air
   
13: SP2(011883510000000091)   goes to iNum echo test and echo test works as expected

The number or url inside the parenthesis show up in the OBi log on the right side beside Peer Number under SP2/Outbound entries in the call log so I think they're being dialed correctly (a DigitMap isn't wonking it)

It's the SIP URIs that are not working.  I'm pretty sure these worked a month or so ago. I don't think I've changed any parameters on the unit that would affect the SIP URI connectivity.  Same router, no firewall changes.  No Stun server on SP2 - never had one on it. Grasping at straws I forwarded ports 5060-5061 to the OBi - still no joy.

What bit have I twiddled incorrectly?

"This has not only been fun, it's been a major expense." - Gallagher

RonR

Quote from: infin8loop on December 06, 2011, 06:23:09 PM
10: SP2(8835100xxxxxxxx@sip.inum.net) fails with End Call (500 Service Unavailable)

Any chance VoIP.ms employs loopback detection and is stopping the call?

Quote from: infin8loop on December 06, 2011, 06:23:09 PM
12: SP2(883510000000091@sip.inum.net)  call connects but just dead air

It could be a router/NAT/RTP problem.  Do you have a STUN server configured in the OBi?

infin8loop

Quote from: RonR on December 06, 2011, 06:39:46 PM
Quote from: infin8loop on December 06, 2011, 06:23:09 PM
10: SP2(8835100xxxxxxxx@sip.inum.net) fails with End Call (500 Service Unavailable)

Any chance VoIP.ms employs loopback detection and is stopping the call?


I don't know for sure but I'll go with loopback detection stopping the call.  I tried a similar speed dial to my iNum at VoxOx and it rang through to the targeted PSTN but still dead air.

Quote from: RonR on December 06, 2011, 06:39:46 PM
Quote from: infin8loop on December 06, 2011, 06:23:09 PM
12: SP2(883510000000091@sip.inum.net)  call connects but just dead air

It could be a router/NAT/RTP problem.  Do you have a STUN server configured in the OBi?


I didn't have a STUN server configured.  I added a STUN server and it still fails.  I also tried forwarding the RTP ports to the OBi and it still fails.  I tried SP(011883510000000091@sip.voip.ms) and that works.  Using similar speed dials formatted as SP2(0118835100xxxxxxxx@sip.voip.ms) with either my voip.ms iNum or VoxOx iNum works. Perhaps this is something awry on the voip.ms side with sip.inum.net.  I'm not up to a chat with voip.ms tech support at the moment.  It's not earth shattering.  Mostly just experimentation.

Thanks for your tips.
"This has not only been fun, it's been a major expense." - Gallagher

RonR

SP2(883510000000091@sip.inum.net) does not go through voip.ms.  The call is made directly between your OBi and sip.inum.net.

infin8loop

Quote from: RonR on December 07, 2011, 11:44:58 AM
SP2(883510000000091@sip.inum.net) does not go through voip.ms.  The call is made directly between your OBi and sip.inum.net.

Well that's good to know for sure. At first I thought that's the way it worked but some how convinced myself maybe it didn't. Glad I didn't involve voip.ms tech support. That being said, I'm not sure where to go from here. My family is already convinced I've lost my mind with the testing I do.  It's probably not long before they stage an OBi Intervention.
"This has not only been fun, it's been a major expense." - Gallagher

RonR


infin8loop

Quote from: RonR on December 07, 2011, 12:02:27 PM
Can you bypass your router momentarily?

Not anytime soon.  I have to wait until The Others leave the house.

I just thought I had no life and then I bought an OBi and now I know it.

   
"This has not only been fun, it's been a major expense." - Gallagher

RonR

BTW,

You misspelled novice:

Run for your lives before this nevice sucks your brain out.

infin8loop

Quote from: RonR on December 07, 2011, 12:28:26 PM
BTW,

You misspelled novice:

Run for your lives before this nevice sucks your brain out.

It's "device" or .... maybe I should be offended.  LOL
"This has not only been fun, it's been a major expense." - Gallagher

RonR


infin8loop

Quote from: RonR on December 07, 2011, 12:02:27 PM
Can you bypass your router momentarily?
I was able to remove the router and connect the OBi110 directly to the cable modem. The SP2(883510000000091@sip.inum.net) speed dial works (with sound) when the router is out of the equation. Using the router, I've tried port forwarding 5060-5061 and 16600-16998 to the OBi and calls connect but no sound.  Neither putting the OBi in the router DMZ nor using a STUN server on the OBi fixes the issue.  When forwarded or in the DMZ the ports are exposed to the outside world per ShieldsUp! port probe at www.grc.com. The router is a first generation wired Netgear RP614 with the most current firmware available dated Dec 2003 (pause to let the laughter stop). Oddly (or maybe not) with the router in place, speed dial SP2(18004444444@sip.tollfreegateway.com) works! 

Observing the OBi Call Status on calls to each of these speed dials I noticed that the RTP stream for calls to sip.inum.net are established on Peer RTP Address 81.201.82.41 while the IP address of sip.inum.net is 81.201.82.25 (different) and they get the no sound issue. On calls to sip.tollfreegateway.com the RTP stream is established on Peer RTP Address 76.10.223.207 and the IP address for sip.tollfreegateway.com is the same, 76.10.223.207 and the calls work with sound. Not sure if this is a coincidence or not.

I swapped a wireless Trendnet TEW-432BRP router in to be the only router (normally it just hangs off the Netgear with DHCP disabled as a wireless AP).  The Trendnet's most currently available firmware is dated Dec 2007 (I know --- more laughter ensues). Going through the same rigmorole on the Trendnet of port forwarding, or DMZ, or STUN server, and it behaves the same way.. no sound on the sip.inum.net calls.

SP2 is still configured for voip.ms and works with "normally" dialed calls (not sip uri). Are these routers functioning like a roach motel with RTP packets checking in and not checking out? With the Netgear router in place I can call the Houston iNum PSTN access number using my cellphone and enter my voip.ms issued iNum and the call comes through on the OBi with sound on both ends. My daughter had her friend in the Netherlands call the Amsterdam iNum PSTN access number and enter the same iNum number and that call rang thorough on the OBi, but when answered, no sound on either end. I didn't catch the Call Status on that call. They were also on a Skype video chat at the time so we could tell what was happening. They were impressed but would have been even MORE impressed had they been actually able to talk to each other on the phone! 

Sinterklass did not bring me a new router nor did I get my initial in dreamy chocolate. I'm considering the Linksys WRT54GL. At this point I really don't need gigabit. Of course the minute I say that, I'll probably need it. Would this router most likely solve my issues? I say "most likely" because I haven't fully ruled out the possibility I have no clue what I'm doing. Yes, perhaps I'm still a "novice" and it's not just the "device" ;-) I'm open to other voip friendly router suggestions.     
"This has not only been fun, it's been a major expense." - Gallagher